The present invention relates generally to audio signal processing, and more particularly to the efficient digital coding and processing of audio signals for use in multimedia and telecommunications applications.
In many multimedia computer applications, both video and audio information are presented simultaneously to the computer user. This simultaneous combination of video and audio information requires a computer to rapidly transfer large quantities of data. That is, the computer must be able to handle high bandwidth data streams. While recent increases in computing power have allowed significant improvements in the multimedia field, the perceived quality of a multimedia presentation can still be limited by the bandwidth capabilities of the host computer system.
Currently, the most commonly used technique for digitally encoding broadband audio information is Pulse Code Modulation, or PCM. In PCM, an analog signal is sampled and converted to a fixed-length binary code. The value of the binary code varies according to the instantaneous sampled amplitude of the analog signal.
The Nyquist sampling theorem establishes the minimum sampling rate (f.sub.s) that can be used with a given analog input signal. For a signal to be reproduced accurately, each cycle of the highest frequency component contained in the analog input signal (f.sub.a) must be sampled at least twice. Consequently, the minimum sampling rate is equal to twice the highest audio input frequency. If f.sub.s is less than 2 times f.sub.a, distortion (known as aliasing or foldover) will result.
Since high fidelity audio signals typically include frequencies up to about 20 kHz, commonly used compact disk (CD) quality audio signal processing systems utilize a sampling rate of approximately 44.1 kHz.
As is known to those skilled in the art, one possible implementation of PCM incorporates signed magnitude, n-bit codes, where n may be any positive whole number greater than 1. The most significant bit (MSB) is the sign bit, and the remaining bits are used to represent magnitude.
The number of PCM bits per sample required to accurately reproduce an audio signal is determined primarily by the distribution of amplitudes present in that audio signal. The greater the distribution of amplitudes, the greater the dynamic range (DR) required of the audio signal processing system. A system's dynamic range (in absolute value) is defined as the ratio of the largest possible signal amplitude that can be encoded by the analog to digital (A/D) converter, to the smallest possible signal amplitude that can be encoded by that A/D converter. In units of decibels (dB), dynamic range equals 20 log(absolute value of DR).
The following mathematical relationship can be used to determine the minimum number of bits required to encode an audio signal of a given dynamic range: EQU 2.sup.n -1=DR
where n equals the number of PCM bits, and DR equals the absolute value of the dynamic range of the audio signal being sampled. By performing the appropriate algebraic manipulations, it can be shown that: EQU n=(log (DR+1))/log2
The distribution of signal amplitude (and thus power) for an audio signal is often found to be inversely proportional to frequency (i.e., proportional to 1/f.sub.a). Consequently, the lower frequency components of an audio signal require a system with a higher dynamic range (more bits per sample) for accurate reproduction, while the higher frequency components of an audio signal can be accurately reproduced with a system having a lower dynamic range (fewer bits per sample).
To accurately reproduce all of the frequency components of an audio signal, typical CD quality audio equipment processes each of two stereo channels using 16 bits per sample, allowing a dynamic range of approximately 96 dB. Since signalling rate (in bits per second) equals the number of bits per sample times the sample rate, the signalling rate required for two 16-bit channels sampled at 44.1 kHz is (2).times.(16).times.(44,100)=1.4112 megabits/second.
Despite the relatively high signalling rate requirements for broadband audio reproduction, current integrated circuit technology is capable of realizing relatively cost effective linear PCM systems. However, such PCM systems are not practical where high quality, low signalling rate audio signal processing is required. PCM is fundamentally inefficient in the encoding of broadband audio signals because the sample rate is determined by the highest frequency to be reproduced, and the number of bits per sample is determined by the lowest range of frequencies to be reproduced. Thus, the signalling rate required by a PCM system to encode broadband audio data is significantly higher than that which would be required if the high and low frequency components were encoded separately.
Accordingly, an object of the present invention is to provide an efficient audio encoding system which separates an analog audio signal into separate bands of frequency components before PCM or other encoding.
It is another object of the present invention to provide relatively high quality, low signalling rate audio signal processing.
Additional objects and advantages of the invention will be set forth in part in the description which follows, and in part become apparent to those skilled in the art upon examination of the following, or may be learned by practice of the invention. The objects and advantages of the invention may be realized and obtained by means of the instrumentalities and combinations particularly pointed out in the claims.